Schedule

Elena-Ramona Modroiu, Co-founder Kamailio, Asipto, Germany

A look at the most important events as well as the features introduced during the past year by the Kamailio project: major releases, with dozens of new modules and what were the engines behind the development of the project.

Alex Balashov, Owner Evariste Systems, USA

A talk about short-duration (“dialler”) traffic in the PSTN. Short-duration calls are unique and challenging; most of them go unanswered, they consume a large volume of resources relative to the revenue they generate, and few vendors want to receive them. Alex will speak to how Kamailio can be used to solve some of these challenges.

Mathias Pasquay, CEO pascom Netzwerktechnik, Germany

A case study on how with the help and support from Kamailio as well as other Open Source projects, we embarked on a project to develop the existing Asterisk based MobyDick PBX to include both Multi-Tennant and Cloud capabilities. All without a complete redesign, expensive hardware and no loss of the numerous benefits of having a customisable on-site PBX.

Simon Woodhead, Simwood, UK

Building a trustable interconnect service it is more than deploying a VoIP routing system. Ability to analyse and proactively interact with customer to prevent fraud as well as ensuring the quality make the big difference, being key components for a successful business. See the evolution over the time of managing a carrier interconnect service, many times facing new challenges, ending up with innovative solutions to move forward.

Andreas Granig, CTO Sipwise, Austria

The time of just deploying a telephony system and then operate it for many years is gone. The market is changing at high speed, user expectation is beyond the voice communication and simple texting. To be able to cope with new demands and even have the flexibility to bring in own ideas, the backbone of a modern telecom has to rely on an open RTC platform. It must control every aspect of it and have the flexibility to adapt quickly in order to meet the customer expectations. Learn from the experiences of interacting with large operators and building such platform at Sipwise, leveraging on open source components such as Kamailio, Asterisk or SEMS

Tim Panton, YoPet.us, UK

A presentation describing some ways developers can use the other capabilities in a browser to enhance a WebRTC user experience. Examples include:

• Using URLs as a rendezvous point
• Using PeerConnection to read QRcodes as a form of password free identity management
• Using WebAudio to inject test tones into a webRTC call
• Using WebAudio to measure Mouth-to-ear latency in a solution
• Using WebAudio to bring smart mixing to a full mesh conference call
• Demos and code walk throughs of each of these examples will be attempted

Alexandr Dubovikov, Founder Homer Sipcapture Project, Germany

SIP Monitoring and Troubleshooting champion HOMER 5 brings a complete redesign of the UI component migrating to a modern framework adding unprecedented flexibility for client side interaction and opening the way for new and exciting functionality extensions powered by Kamailio’s sipcapture module’s leading HEP3 support handling inbound packets, statistics and arbitrary logs. Trusty companion CAPTAGENT 5 completely new internal architecture and expands its performance range and core capabilities to handle more network protocols, relay traffic for remote agents, parse and handle RTCP and RTP statistics, RPC centralized control and much more.

Sebastian Damm, VoIP Architect, sipgate, Germany

A talk focusing on how to smoothen the process to enable communication between existing IPv4 SIP-VoIP deployments and IPv6 servers or endpoints, how can Kamailio help to achieve such goals and where you have to pay attention.

Olle E. Johansson, Owner Edvina, Sweden

Is your real time communication service ready for the 21st century?

Matthew Jordan, Lead of Asterisk PBX Project, Digium, USA

Asterisk is often deployed to fulfill a variety of roles. Often, a single instance of Asterisk will be dedicated to some task – be it call recording, call queuing, voicemail, IVRs, or some combination thereof. This method can work for many deployments, but it can also be difficult to scale. As customer needs for a particular resource increases, existing Asterisk instances cannot easily be re-purposed to fulfill that use case. This can result in inefficient utilization of server resources and increased complexity of system maintenance.This talk will demonstrate how Asterisk can be treated as a generic media application server, making use of the new Asterisk REST Interface (ARI). Along with Kamailio and other technologies, this allows for a highly scalable communications architecture.

Seudin Kasumovic, Bicom Systems

Erlang is one of the famous programming languages in telephony. Next Kamailio release will have a dedicated module to interact with Erlang nodes, enabling RPC calls to Erlang nodes directly from the Kamailio configuration file and exporting Kamailio’s RPC commands to Erlang. Beyond that, an extensive developer APIs allow the creation of custom modules for specific applications. The talk will go trough the benefits of Erlang, the new module capabilities and it will show several how-to examples.

Sherman Scholten, Obihai, USA

Zero-Touch / Light-Touch Service Activation and Out of Band Cloud-Management Techniques for IP Phone and Phone Adapter CPE – a modern core platform is not sufficient to build a successful telephony services. Ability to interact with end user equipment is essential for smooth operations as well as better user experience. The presentation will show how Obihai addressed the expectation of end users for telephony terminals and enabled operators to have full control of devices with flexible APIs.

Dennis P. Kersten, Germany

We live in a world defined by communication, that is most probably the reason why we all here are in this industry. Still our everyday business is not about interacting and creating new things with other RTC geeks, can we improve something in this regard?. The talk will present about the VoIP Lab Berlin initiative, how it can help everyone in open source and telecom industry, revealing already its first secret fruits.

Randy Resnick, Founder VoIP Users Conference, France

What is new and exciting in the real time communications? What will be there in one or two years from now? Open discussions with a selected group of guests.

Among Panelists: Alan Duric (CTO Wire app), Simon Tennant (Founder Buddycloud and XMPP Software Foundation member), Tim Panton (WebRTC and Protocol Droid at Yopet.us)

Dan Christian Bogos, Co-founder CGRateS Project, Germany

In this talk Dan will introduce to the audience the LCR component of CGRateS, adding dynamic behavior to LCR rules through internal mechanisms like real-time QoS metrics or Account Thresholds with automatic Triggers manipulating routing tables. CGRateS is a battle-tested real-time Charging Engine which has recently integrated Kamailio support for all of it’s charging modes: Prepaid, Pseudo-prepaid, Postpaid and Rated.

Carsten Bock, CEO NG Voice, Germany

Voice communication is only one aspect of VoLTE, given that legacy SMS-Service is still mandatory for most networks. Implementing a proper SMS-Service is quite often a challenge, due to the fact, that SMS is a relict of the 2G/3G era. Kannel.org offers various interfaces for SMS, while it still is missing a SIP Interface. Kamailio is great for SIP with support for Instant Messaging, but has very limited native SMS functionality. This talk will show how to interconnect the two applications to bridge text messages: Kamailio as an Application-Server for SIP and Kannel as a gateway to other transmission networks.

Nir Simionovich, Owner Greenfield Technologies, Israel

SBC’s, Soft Switched, Media Servers, Applications Servers, Core IMS, etc – these are the things that make our VoIP world turn around. But have they truly solved the issue of platform scalability? Do they truly provide a means of building large scale platforms? What does it take to build something like Skype? Can it be done using Open Source code? The answer is in Federated BoIP Switching, a new methodology of thinking about VoIP switching, SIP registration and how they all combine.

Matthew Hodgson, Technical Co-Founder, Matrix.org, UK

Real-time communication on the current internet is alarmingly fragmented – between apps like WhatsApp, Viber, etc, and hundreds of WebRTC startups there is almost zero interoperability. The users have lost control over ownership of their communications. Matrix is an open source initiative to solve this problem, with the ambitious goal of creating a world in which calling or messaging a contact is as simple as emailing them. Matrix is the missing application layer on top of WebRTC for open interoperable real-time communication on the internet. This session will introduce Matrix, its mission, its open standard and open source reference implementations, and their relevance to the VoIP and WebRTC communities.

Thomas Magedanz, Prof. Dr., FhG Fokus Institute, Germany

Addressing the prototyping challenges of emerging 5G Standards, Platforms and Applications

There is no doubt that SIP and IMS will play their role in future multimedia communications on top of fixed broadband and LTE and beyond. However, looking at the impacts of NFV and SDN concepts on the evolution of IMS, it becomes clear that virtualised IMS solutions and simple web clients accessing them from any end system for human to human communications are becoming the new standard. Looking further on the drivers of communications, the emerging internet of things and the related M2M communications will become economically very relevant. This becomes clear when looking at the identified 5G applications domains, which are mainly industrial IOT applications. Looking at the NGMN White Paper 5G Reference Architecture, key will be the SDN/NFV-based dynamic provision of application specific overlay network and service platforms, and thus voice / multimedia overlays will be hosted and operated next to IOT and content delivery network slices. In order to test and validate new architectures and perform adequate benchmarking of those new architectures and applications seamlessly on top of heterogeneous access networks, the new FOKUS Open5G Playground provides to its partners and customers a unique technical 5G testbed infrastructure based on the Open5GMTC, OpenSDNCore and Open5GCore toolkits integrating various access network technologies. Also new Voice and MM overlay architectures can be tested.

James Body, Head of Research and Development, Truphone, UK

Interactive session:

• Individuals give ‘Dangerous Demos’ of subjects containing material that is exciting, educational, entertaining, energetic and potentially explosive.
• Slots strictly limited to 4 minutes
• Three prizes:
  • Most Entertaining Demo (selected by judges/panel)
  • Audience Choice (selected using live (dangerous) voting system)
  • Most Ridiculously Risky Demo (aka the demo that crashes and burns most spectacularly)

Sergio García Ramos, Quobis, Spain

Bluebox-ng is an Open Source VoIP/UC vulnerability scanner. This presentations is going through some of the most common security problems in VoIP/UC systems and how to detect them using Bluebox-ng. Finally it will show how this tool can be used to implement an automatic system to continuously test the VoIP infrastructure (Continuous Pentesting), to be safe that updates to Kamailio, provisioning data or other VoIP applications in the platform are not exposing new security risks.

Federico Cabiddu, Libon Project – Orange, France

Mobile oriented services present new challenges with respect to traditional SIP services. In particular, use of push notifications change the way we are used to think location and registration. We’ll see how tsilo extension along with other Kamailio modules can help to face the challenges, keeping the Kamailio’s well known flexibility.

Luis Azedo, 2600hz, USA

A talk about the general purpose AMQP connector that has been completely rewritten for Kamailio 4.2 to interact with Kazoo cloud telephony platform. This module provides integration into your program, instead of your database, so you can overlay additional logic in your preferred language while also utilizing a distributed messaging layer. The 2600hz Kazoo platform uses this capability to create loosely-connected, highly-cohesive SIP servers which interact in real-time with application logic. Learn how to harness the power of Kazoo or the Kazoo Kamailio module directly.

Tudor Golubenco, Lead Developer, Packetbeat Project

We already have good tools for monitoring and troubleshooting SIP calls and transactions. But quite often the performance and availability issues are coming not from Kamailio itself but rather from the services that Kamailio queries, like MySQL, DNS or RADIUS.Packetbeat is an open source application tracing tool based on Elasticsearch. The talk will present several ideas of using it to trace the performance of Kamailio-based systems and to measure the interest in such approach.

Dragos Vingarzan, CTO Core Network Dynamics, Germany

The time when only big operators could afford to do it is gone, companies or special entities can deploy their own small private network. The talk will focus on the technologies to deploy 4G networks with compatibility for older versions 3G and 2G, offering the the best possible services based on signal coverage.

Jose Luis Millan, Co-Founder JsSIP Project, Spain

JsSIP is a state of the art SIP library implementation in JavaScript. The talk is focusing on showing how it can be used to built WebRTC SIP applications with just few lines of JavaScript code and HTML, offering Voice, Video and Instant Messaging. Using modern browsers, anyone can call from a web page via your Kamailio SIP Server with the websocket module enabled, opening your existing set of services to a new world as well as enriching your services with contextual communications.

Stefan Sayer, Lead Developer SEMS Project, Germany

A fast walk through what is new in SIP Express Media Server

Daniel-Constantin Mierla, Co-founder Kamailio, Asipto, Germany

Up to date tutorial about the tools provided nowadays by Kamailio to secure your SIP-VoIP service. Encryption, DoS and DDoS mitigation, dictionary attacks protection and anti-fraud mechanisms.